H.323 is an umbrella recommendation from the International Telecommunications Union (ITU) that set standards for multimedia communications over packet switched network that do not provide a guaranteed Quality of Service. Such networks are pervasive on many corporate terminals and include TCP/IP and IPX over Ethernet, Fast Ethernet and Token Ring network technologies. The H.323 standard, titled: Packet-Based Multimedia Communications Systems, provides a foundation for audio, video, and data communications across IP-based networks, including the Internet. Multimedia products and applications complying with the H.323 standard are interoperable, can communicate with each other, and thus are compatible. Many sub standards make up the H.323 standard or protocol, one of which is the H.245 standard. The term real-time multimedia communication used herein is addressed to communication according to this standard, but the present invention can also be used in relation with other similar standards and protocols, e.g. the SIP (Session Initiated Protocol) standard.
Transmitting data by a packet-switched network is one of the most common methods of transmitting data. As with any other type of data transmission, data transmitted by a packet-switched network can be affected by transmission errors such as loss of packets. Real-time multimedia communication is particularly exposed to packet loss because there is usually no time for error correction communication between the parties as can be done in other types of data transmission. A packet loss of only a few percentages will e.g. occur as annoying disturbance and interruptions on the video display of the receiver.
Besides, due to the video coding characteristics of e.g. H.323 multimedia communication, loss of some packets in a data flow can significantly affect the rest of the data flow. For example, consider the situation where the transmitted data is digital video encoded by a prediction based compression technique. In that case, loss of packets will affect not only a particular frame to which the data in the lost packets belong, but also subsequent frames. In addition, if the compression technique uses motion compensation, then the lost packets will affect not only a particular region in the frames, but also surrounding regions in the subsequent frames, the extent of which depends on the value of the motion vectors. Similarly, if the transmitted data is digital video encoded using variable length coding (for example, Huffman coding), the packet loss can render the information contained in one or more of the subsequent packets unusable. Various techniques have been developed to minimize, and even correct for, the effects of packet loss on transmitted video data. One set of techniques attempt to reduce the effect of packet loss by including redundant control data in all packets. For example, some packetization protocols require control data necessary for decoding a packet to be included in a packet's header, even though the same information is included in a preceding packet.
Another set of techniques attempt to reduce the effects of lost data on the video image by replacing the lost data with other data. For example, according to one such technique, the lost data is replaced with data from a preceding frame, thereby attempting to improve the image quality of the current frame and reduce errors in subsequently decoded frames. Yet another set of techniques provide methodologies for allowing a receiving terminal to determine whether a packet has been lost and, if so, send a request for a correction of the lost data to the transmitting terminal. The transmitting terminal then provides data, which corrects the effects of the lost data.
The attempt to reduce the effects of packet loss mentioned above is only useful in situations of marginal or momentary packet loss. In many cases, packet loss occurs e.g. when the terminals try to communicate with a higher bit rate than the assigned transmission pipe allows. As an example, when a user requests a call on 768 kbps on a 768 kbps communication link, the link will not be able to transfer as much media data as requested, due to signalling and overhead. The difference between requested and actual media data throughput will occur as packet loss implying annoying disturbance on the user's video display.
According to prior art, the problems of packet loss occurring when data of too high bit rate relative to the capacity of the communication pipe is transmitted are solved by so-called down speeding. Down speeding means stepwise data rate reduction, of which data is transmitted from the multimedia terminals during a call.
According to the H.245 standard, which defines the control protocol part of the H.323 standard, there are several ways of altering the flow rate during a call. One way is to send a flowControlCommand to the transmitting terminal. The flowControlCommand includes the fields logicalChannelNumber and maximumBitRate. The maximumBitRate parameter then indicates the maximum allowed bit rate for the logical channel, to which the transmission data rate is reduced.